src/java.desktop/windows/native/libjsound/PLATFORM_API_WinOS_DirectSound.cpp - master - wstartsev/jdk

src/java.desktop/windows/native/libjsound/PLATFORM_API_WinOS_DirectSound.cpp
1 443 строки50 KB
Magnus Ihse Bursie
8339783: Implement JEP 479: Remove the Windows 32-bit x86 Port
13 ноя 2024, 12:41
13 ноя 2024, 12:4179345bb
/*
 * Copyright (c) 2003, 2024, Oracle and/or its affiliates. All rights reserved.
 * DO NOT ALTER OR REMOVE COPYRIGHT NOTICES OR THIS FILE HEADER.
 *
 * This code is free software; you can redistribute it and/or modify it
 * under the terms of the GNU General Public License version 2 only, as
 * published by the Free Software Foundation.  Oracle designates this
 * particular file as subject to the "Classpath" exception as provided
 * by Oracle in the LICENSE file that accompanied this code.
 *
 * This code is distributed in the hope that it will be useful, but WITHOUT
 * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
 * FITNESS FOR A PARTICULAR PURPOSE.  See the GNU General Public License
 * version 2 for more details (a copy is included in the LICENSE file that
 * accompanied this code).
 *
 * You should have received a copy of the GNU General Public License version
 * 2 along with this work; if not, write to the Free Software Foundation,
 * Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA.
 *
 * Please contact Oracle, 500 Oracle Parkway, Redwood Shores, CA 94065 USA
 * or visit www.oracle.com if you need additional information or have any
 * questions.
 */

#define USE_ERROR
#define USE_TRACE

/* define this for the silencing/servicing code. Requires USE_TRACE */
//#define USE_DEBUG_SILENCING

#ifndef WIN32_EXTRA_LEAN
#define WIN32_EXTRA_LEAN
#endif
#ifndef WIN32_LEAN_AND_MEAN
#define WIN32_LEAN_AND_MEAN
#endif

#include <windows.h>
#include <mmsystem.h>
#include <string.h>

/* include DirectSound headers */
#include <dsound.h>

/* include Java Sound specific headers as C code */
#ifdef __cplusplus
extern "C" {
#endif
 #include "DirectAudio.h"
#ifdef __cplusplus
}
#endif

/* include to prevent charset problem */
#include "PLATFORM_API_WinOS_Charset_Util.h"

#ifdef USE_DEBUG_SILENCING
#define DEBUG_SILENCING0(p) TRACE0(p)
#define DEBUG_SILENCING1(p1,p2) TRACE1(p1,p2)
#define DEBUG_SILENCING2(p1,p2,p3) TRACE2(p1,p2,p3)
#else
#define DEBUG_SILENCING0(p)
#define DEBUG_SILENCING1(p1,p2)
#define DEBUG_SILENCING2(p1,p2,p3)
#endif


#if USE_DAUDIO == TRUE

/* 3 seconds to wait before device list is re-read */
#define WAIT_BETWEEN_CACHE_REFRESH_MILLIS 3000

/* maximum number of supported devices, playback+capture */
#define MAX_DS_DEVICES 60

typedef struct {
    INT32 mixerIndex;
    BOOL isSource;
    /* either LPDIRECTSOUND or LPDIRECTSOUNDCAPTURE */
    void* dev;
    /* how many instances use the dev */
    INT32 refCount;
    GUID guid;
} DS_AudioDeviceCache;

static DS_AudioDeviceCache g_audioDeviceCache[MAX_DS_DEVICES];
static INT32 g_cacheCount = 0;
static UINT64 g_lastCacheRefreshTime = 0;
static INT32 g_mixerCount = 0;

BOOL DS_lockCache() {
    /* dummy implementation for now, Java does locking */
    return TRUE;
}

void DS_unlockCache() {
    /* dummy implementation for now */
}

static GUID CLSID_DAUDIO_Zero = {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0};

BOOL isEqualGUID(LPGUID lpGuid1, LPGUID lpGuid2) {
    if (lpGuid1 == NULL || lpGuid2 == NULL) {
        if (lpGuid1 == lpGuid2) {
            return TRUE;
        }
        if (lpGuid1 == NULL) {
            lpGuid1 = (LPGUID) (&CLSID_DAUDIO_Zero);
        } else {
            lpGuid2 = (LPGUID) (&CLSID_DAUDIO_Zero);
        }
    }
    return memcmp(lpGuid1, lpGuid2, sizeof(GUID)) == 0;
}

INT32 findCacheItemByGUID(LPGUID lpGuid, BOOL isSource) {
    int i;
    for (i = 0; i < g_cacheCount; i++) {
        if (isSource == g_audioDeviceCache[i].isSource
            && isEqualGUID(lpGuid, &(g_audioDeviceCache[i].guid))) {
            return i;
        }
    }
    return -1;
}

INT32 findCacheItemByMixerIndex(INT32 mixerIndex) {
    int i;
    for (i = 0; i < g_cacheCount; i++) {
        if (g_audioDeviceCache[i].mixerIndex == mixerIndex) {
            return i;
        }
    }
    return -1;
}

typedef struct {
    INT32 currMixerIndex;
    BOOL isSource;
} DS_RefreshCacheStruct;


BOOL CALLBACK DS_RefreshCacheEnum(LPGUID lpGuid,
                                  LPCSTR lpstrDescription,
                                  LPCSTR lpstrModule,
                                  DS_RefreshCacheStruct* rs) {
    INT32 cacheIndex = findCacheItemByGUID(lpGuid, rs->isSource);
    /*TRACE3("Enumerating %d: %s (%s)\n", cacheIndex, lpstrDescription, lpstrModule);*/
    if (cacheIndex == -1) {
        /* add this device */
        if (g_cacheCount < MAX_DS_DEVICES-1) {
            g_audioDeviceCache[g_cacheCount].mixerIndex = rs->currMixerIndex;
            g_audioDeviceCache[g_cacheCount].isSource = rs->isSource;
            g_audioDeviceCache[g_cacheCount].dev = NULL;
            g_audioDeviceCache[g_cacheCount].refCount = 0;
            if (lpGuid == NULL) {
                memset(&(g_audioDeviceCache[g_cacheCount].guid), 0, sizeof(GUID));
            } else {
                memcpy(&(g_audioDeviceCache[g_cacheCount].guid), lpGuid, sizeof(GUID));
            }
            g_cacheCount++;
            rs->currMixerIndex++;
        } else {
            /* failure case: more than MAX_DS_DEVICES available... */
        }
    } else {
        /* device already exists in cache... update mixer number */
        g_audioDeviceCache[cacheIndex].mixerIndex = rs->currMixerIndex;
        rs->currMixerIndex++;
    }
    /* continue enumeration */
    return TRUE;
}

///// implemented functions of DirectAudio.h

INT32 DAUDIO_GetDirectAudioDeviceCount() {
    DS_RefreshCacheStruct rs;
    INT32 oldCount;
    INT32 cacheIndex;

    if (!DS_lockCache()) {
        return 0;
    }

    HRESULT hr = ::CoInitializeEx(NULL, COINIT_MULTITHREADED | COINIT_DISABLE_OLE1DDE);
    if (FAILED(hr) && hr != RPC_E_CHANGED_MODE) {
        DS_unlockCache();
        return 0;
    }

    if (g_lastCacheRefreshTime == 0
        || (UINT64) timeGetTime() > (UINT64) (g_lastCacheRefreshTime + WAIT_BETWEEN_CACHE_REFRESH_MILLIS)) {
        /* first, initialize any old cache items */
        for (cacheIndex = 0; cacheIndex < g_cacheCount; cacheIndex++) {
            g_audioDeviceCache[cacheIndex].mixerIndex = -1;
        }

        /* enumerate all devices and either add them to the device cache,
         * or refresh the mixer number
         */
        rs.currMixerIndex = 0;
        rs.isSource = TRUE;
        DirectSoundEnumerate((LPDSENUMCALLBACK) DS_RefreshCacheEnum, &rs);
        /* if we only got the Primary Sound Driver (GUID=NULL),
         * then there aren't any playback devices installed */
        if (rs.currMixerIndex == 1) {
            cacheIndex = findCacheItemByGUID(NULL, TRUE);
            if (cacheIndex == 0) {
                rs.currMixerIndex = 0;
                g_audioDeviceCache[0].mixerIndex = -1;
                TRACE0("Removing stale Primary Sound Driver from list.\n");
            }
        }
        oldCount = rs.currMixerIndex;
        rs.isSource = FALSE;
        DirectSoundCaptureEnumerate((LPDSENUMCALLBACK) DS_RefreshCacheEnum, &rs);
        /* if we only got the Primary Sound Capture Driver (GUID=NULL),
         * then there aren't any capture devices installed */
        if ((rs.currMixerIndex - oldCount) == 1) {
            cacheIndex = findCacheItemByGUID(NULL, FALSE);
            if (cacheIndex != -1) {
                rs.currMixerIndex = oldCount;
                g_audioDeviceCache[cacheIndex].mixerIndex = -1;
                TRACE0("Removing stale Primary Sound Capture Driver from list.\n");
            }
        }
        g_mixerCount = rs.currMixerIndex;

        g_lastCacheRefreshTime = (UINT64) timeGetTime();
    }

    if (hr != RPC_E_CHANGED_MODE) {
        ::CoUninitialize();
    }

    DS_unlockCache();
    /*TRACE1("DirectSound: %d installed devices\n", g_mixerCount);*/
    return g_mixerCount;
}

BOOL CALLBACK DS_GetDescEnum(LPGUID lpGuid,
                             LPCWSTR lpstrDescription,
                             LPCWSTR lpstrModule,
                             DirectAudioDeviceDescription* desc) {

    INT32 cacheIndex = findCacheItemByGUID(lpGuid, g_audioDeviceCache[desc->deviceID].isSource);
    if (cacheIndex == desc->deviceID) {
        UnicodeToUTF8AndCopy(desc->name, lpstrDescription, DAUDIO_STRING_LENGTH);
        //strncpy(desc->description, lpstrModule, DAUDIO_STRING_LENGTH);
        desc->maxSimulLines = -1;
        /* do not continue enumeration */
        return FALSE;
    }
    return TRUE;
}


INT32 DAUDIO_GetDirectAudioDeviceDescription(INT32 mixerIndex, DirectAudioDeviceDescription* desc) {

    if (!DS_lockCache()) {
        return FALSE;
    }

    /* set the deviceID field to the cache index */
    desc->deviceID = findCacheItemByMixerIndex(mixerIndex);
    if (desc->deviceID < 0) {
        DS_unlockCache();
        return FALSE;
    }

    HRESULT hr = ::CoInitializeEx(NULL, COINIT_MULTITHREADED | COINIT_DISABLE_OLE1DDE);
    if (FAILED(hr) && hr != RPC_E_CHANGED_MODE) {
        DS_unlockCache();
        return 0;
    }

    desc->maxSimulLines = 0;
    if (g_audioDeviceCache[desc->deviceID].isSource) {
        DirectSoundEnumerateW((LPDSENUMCALLBACKW) DS_GetDescEnum, desc);
        strncpy(desc->description, "DirectSound Playback", DAUDIO_STRING_LENGTH);
    } else {
        DirectSoundCaptureEnumerateW((LPDSENUMCALLBACKW) DS_GetDescEnum, desc);
        strncpy(desc->description, "DirectSound Capture", DAUDIO_STRING_LENGTH);
    }

    if (hr != RPC_E_CHANGED_MODE) {
        ::CoUninitialize();
    }

    /*desc->vendor;
    desc->version;*/

    DS_unlockCache();
    return (desc->maxSimulLines == -1)?TRUE:FALSE;
}

/* multi-channel info: http://www.microsoft.com/whdc/hwdev/tech/audio/multichaud.mspx */

//static UINT32 sampleRateArray[] = { 8000, 11025, 16000, 22050, 32000, 44100, 48000, 56000, 88000, 96000, 172000, 192000 };
static INT32 sampleRateArray[] = { -1 };
static INT32 channelsArray[] = { 1, 2};
static INT32 bitsArray[] = { 8, 16};

#define SAMPLERATE_COUNT sizeof(sampleRateArray)/sizeof(INT32)
#define CHANNELS_COUNT sizeof(channelsArray)/sizeof(INT32)
#define BITS_COUNT sizeof(bitsArray)/sizeof(INT32)

void DAUDIO_GetFormats(INT32 mixerIndex, INT32 deviceID, int isSource, void* creator) {

    int rateIndex, channelIndex, bitIndex;

    /* no need to lock, since deviceID identifies the device sufficiently */

    /* sanity */
    if (deviceID >= g_cacheCount) {
        return;
    }
    if ((g_audioDeviceCache[deviceID].isSource && !isSource)
        || (!g_audioDeviceCache[deviceID].isSource && isSource)) {
        /* only support Playback or Capture */
        return;
    }

    for (rateIndex = 0; rateIndex < SAMPLERATE_COUNT; rateIndex++) {
        for (channelIndex = 0; channelIndex < CHANNELS_COUNT; channelIndex++) {
            for (bitIndex = 0; bitIndex < BITS_COUNT; bitIndex++) {
                DAUDIO_AddAudioFormat(creator, bitsArray[bitIndex],
                                      ((bitsArray[bitIndex] + 7) / 8) * channelsArray[channelIndex],
                                      channelsArray[channelIndex],
                                      (float) sampleRateArray[rateIndex],
                                      DAUDIO_PCM,
                                      (bitsArray[bitIndex]==8)?FALSE:TRUE,  /* signed */
                                      (bitsArray[bitIndex]==8)?FALSE:
#ifndef _LITTLE_ENDIAN
                                      TRUE /* big endian */
#else
                                      FALSE /* little endian */
#endif
                                      );
            }
        }
    }
}

typedef struct {
    int deviceID;
    /* for convenience */
    BOOL isSource;
    /* the secondary buffer (Playback) */
    LPDIRECTSOUNDBUFFER playBuffer;
    /* the secondary buffer (Capture) */
    LPDIRECTSOUNDCAPTUREBUFFER captureBuffer;

    /* size of the directsound buffer, usually 2 seconds */
    int dsBufferSizeInBytes;

    /* size of the read/write-ahead, as specified by Java */
    int bufferSizeInBytes;
    int bitsPerSample;
    int frameSize; // storage size in Bytes

    UINT64 framePos;
    /* where to write into the buffer.
     * -1 if at current position (Playback)
     * For Capture, this is the read position
     */
    int writePos;

    /* if start() had been called */
    BOOL started;

    /* how many bytes there is silence from current write position */
    int silencedBytes;

    BOOL underrun;

} DS_Info;


LPCSTR TranslateDSError(HRESULT hr) {
    switch(hr) {
        case DSERR_ALLOCATED:
            return "DSERR_ALLOCATED";

        case DSERR_CONTROLUNAVAIL:
            return "DSERR_CONTROLUNAVAIL";

        case DSERR_INVALIDPARAM:
            return "DSERR_INVALIDPARAM";

        case DSERR_INVALIDCALL:
            return "DSERR_INVALIDCALL";

        case DSERR_GENERIC:
            return "DSERR_GENERIC";

        case DSERR_PRIOLEVELNEEDED:
            return "DSERR_PRIOLEVELNEEDED";

        case DSERR_OUTOFMEMORY:
            return "DSERR_OUTOFMEMORY";

        case DSERR_BADFORMAT:
            return "DSERR_BADFORMAT";

        case DSERR_UNSUPPORTED:
            return "DSERR_UNSUPPORTED";

        case DSERR_NODRIVER:
            return "DSERR_NODRIVER";

        case DSERR_ALREADYINITIALIZED:
            return "DSERR_ALREADYINITIALIZED";

        case DSERR_NOAGGREGATION:
            return "DSERR_NOAGGREGATION";

        case DSERR_BUFFERLOST:
            return "DSERR_BUFFERLOST";

        case DSERR_OTHERAPPHASPRIO:
            return "DSERR_OTHERAPPHASPRIO";

        case DSERR_UNINITIALIZED:
            return "DSERR_UNINITIALIZED";

        default:
            return "Unknown HRESULT";
        }
}

/*
** data/routines for starting DS buffers by separate thread
** (joint into DS_StartBufferHelper class)
** see cr6372428: playback fails after exiting from thread that has started it
** due IDirectSoundBuffer8::Play() description:
** http://msdn.microsoft.com/archive/default.asp?url=/archive/en-us/directx9_c
**       /directx/htm/idirectsoundbuffer8play.asp
** (remark section): If the application is multithreaded, the thread that plays
** the buffer must continue to exist as long as the buffer is playing.
** Buffers created on WDM drivers stop playing when the thread is terminated.
** IDirectSoundCaptureBuffer8::Start() has the same remark:
** http://msdn.microsoft.com/archive/default.asp?url=/archive/en-us/directx9_c
**       /directx/htm/idirectsoundcapturebuffer8start.asp
*/
class DS_StartBufferHelper {
public:
    /* starts DirectSound buffer (playback or capture) */
    static HRESULT StartBuffer(DS_Info* info);
    /* checks for initialization success */
    static inline BOOL isInitialized() { return data.threadHandle != NULL; }
protected:
    DS_StartBufferHelper() {}  // no need to create an instance

    /* data class */
    class Data {
    public:
        Data();
        ~Data();
        // public data to access from parent class
        CRITICAL_SECTION crit_sect;
        volatile HANDLE threadHandle;
        volatile HANDLE startEvent;
        volatile HANDLE startedEvent;
        volatile DS_Info* line2Start;
        volatile HRESULT startResult;
    } static data;

    /* StartThread function */
    static DWORD WINAPI ThreadProc(void *param);
};

/* StartBufferHelper class implementation
*/
DS_StartBufferHelper::Data DS_StartBufferHelper::data;

DS_StartBufferHelper::Data::Data() {
    threadHandle = NULL;
    ::InitializeCriticalSection(&crit_sect);
    startEvent = ::CreateEvent(NULL, FALSE, FALSE, NULL);
    startedEvent = ::CreateEvent(NULL, FALSE, FALSE, NULL);
    if (startEvent != NULL && startedEvent != NULL)
        threadHandle = ::CreateThread(NULL, 0, ThreadProc, NULL, 0, NULL);
}

DS_StartBufferHelper::Data::~Data() {
    ::EnterCriticalSection(&crit_sect);
    if (threadHandle != NULL) {
        // terminate thread
        line2Start = NULL;
        ::SetEvent(startEvent);
        ::CloseHandle(threadHandle);
        threadHandle = NULL;
    }
    ::LeaveCriticalSection(&crit_sect);
    // won't delete startEvent/startedEvent/crit_sect
    // - Windows will do during process shutdown
}

DWORD WINAPI DS_StartBufferHelper::ThreadProc(void *param)
{
    ::CoInitialize(NULL);
    while (1) {
        // wait for something to do
        ::WaitForSingleObject(data.startEvent, INFINITE);
        if (data.line2Start == NULL) {
            // (data.line2Start == NULL) is a signal to terminate thread
            break;
        }
        if (data.line2Start->isSource) {
            data.startResult =
                data.line2Start->playBuffer->Play(0, 0, DSBPLAY_LOOPING);
        } else {
            data.startResult =
                data.line2Start->captureBuffer->Start(DSCBSTART_LOOPING);
        }
        ::SetEvent(data.startedEvent);
    }
    ::CoUninitialize();
    return 0;
}

HRESULT DS_StartBufferHelper::StartBuffer(DS_Info* info) {
    HRESULT hr;
    ::EnterCriticalSection(&data.crit_sect);
    if (!isInitialized()) {
        ::LeaveCriticalSection(&data.crit_sect);
        return E_FAIL;
    }
    data.line2Start = info;
    ::SetEvent(data.startEvent);
    ::WaitForSingleObject(data.startedEvent, INFINITE);
    hr = data.startResult;
    ::LeaveCriticalSection(&data.crit_sect);
    return hr;
}


/* helper routines for DS buffer positions */
/* returns distance from pos1 to pos2
 */
inline int DS_getDistance(DS_Info* info, int pos1, int pos2) {
    int distance = pos2 - pos1;
    while (distance < 0)
        distance += info->dsBufferSizeInBytes;
    return distance;
}

/* adds 2 positions
 */
inline int DS_addPos(DS_Info* info, int pos1, int pos2) {
    int result = pos1 + pos2;
    while (result >= info->dsBufferSizeInBytes)
        result -= info->dsBufferSizeInBytes;
    return result;
}


BOOL DS_addDeviceRef(INT32 deviceID) {
    HWND ownerWindow;
    HRESULT res = DS_OK;
    LPDIRECTSOUND devPlay;
    LPDIRECTSOUNDCAPTURE devCapture;
    LPGUID lpGuid = NULL;


    if (g_audioDeviceCache[deviceID].dev == NULL) {
        /* Create DirectSound */
        TRACE1("Creating DirectSound object for device %d\n", deviceID);
        lpGuid = &(g_audioDeviceCache[deviceID].guid);
        if (isEqualGUID(lpGuid, NULL)) {
            lpGuid = NULL;
        }
        if (g_audioDeviceCache[deviceID].isSource) {
            res = DirectSoundCreate(lpGuid, &devPlay, NULL);
            g_audioDeviceCache[deviceID].dev = (void*) devPlay;
        } else {
            res = DirectSoundCaptureCreate(lpGuid, &devCapture, NULL);
            g_audioDeviceCache[deviceID].dev = (void*) devCapture;
        }
        g_audioDeviceCache[deviceID].refCount = 0;
        if (FAILED(res)) {
            ERROR1("DAUDIO_Open: ERROR: Failed to create DirectSound: %s", TranslateDSError(res));
            g_audioDeviceCache[deviceID].dev = NULL;
            return FALSE;
        }
        if (g_audioDeviceCache[deviceID].isSource) {
            ownerWindow = GetForegroundWindow();
            if (ownerWindow == NULL) {
                ownerWindow = GetDesktopWindow();
            }
            TRACE0("DAUDIO_Open: Setting cooperative level\n");
            res = devPlay->SetCooperativeLevel(ownerWindow, DSSCL_NORMAL);
            if (FAILED(res)) {
                ERROR1("DAUDIO_Open: ERROR: Failed to set cooperative level: %s", TranslateDSError(res));
                return FALSE;
            }
        }
    }
    g_audioDeviceCache[deviceID].refCount++;
    return TRUE;
}

#define DEV_PLAY(devID)    ((LPDIRECTSOUND) g_audioDeviceCache[devID].dev)
#define DEV_CAPTURE(devID) ((LPDIRECTSOUNDCAPTURE) g_audioDeviceCache[devID].dev)

void DS_removeDeviceRef(INT32 deviceID) {

    if (g_audioDeviceCache[deviceID].refCount) {
        g_audioDeviceCache[deviceID].refCount--;
    }
    if (g_audioDeviceCache[deviceID].refCount == 0) {
        if (g_audioDeviceCache[deviceID].dev != NULL) {
            if (g_audioDeviceCache[deviceID].isSource) {
                DEV_PLAY(deviceID)->Release();
            } else {
                DEV_CAPTURE(deviceID)->Release();
            }
            g_audioDeviceCache[deviceID].dev = NULL;
        }
    }
}

#ifndef _WAVEFORMATEXTENSIBLE_
#define _WAVEFORMATEXTENSIBLE_
typedef struct {
    WAVEFORMATEX    Format;
    union {
        WORD wValidBitsPerSample;       /* bits of precision  */
        WORD wSamplesPerBlock;          /* valid if wBitsPerSample==0 */
        WORD wReserved;                 /* If neither applies, set to zero. */
    } Samples;
    DWORD           dwChannelMask;      /* which channels are */
                                        /* present in stream  */
    GUID            SubFormat;
} WAVEFORMATEXTENSIBLE, *PWAVEFORMATEXTENSIBLE;
#endif // !_WAVEFORMATEXTENSIBLE_

#if !defined(WAVE_FORMAT_EXTENSIBLE)
#define  WAVE_FORMAT_EXTENSIBLE                 0xFFFE
#endif // !defined(WAVE_FORMAT_EXTENSIBLE)

#if !defined(DEFINE_WAVEFORMATEX_GUID)
#define DEFINE_WAVEFORMATEX_GUID(x) (USHORT)(x), 0x0000, 0x0010, 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71
#endif
#ifndef STATIC_KSDATAFORMAT_SUBTYPE_PCM
#define STATIC_KSDATAFORMAT_SUBTYPE_PCM\
    DEFINE_WAVEFORMATEX_GUID(WAVE_FORMAT_PCM)
#endif


void createWaveFormat(WAVEFORMATEXTENSIBLE* format,
                      int sampleRate,
                      int channels,
                      int bits,
                      int significantBits) {
    GUID subtypePCM = {STATIC_KSDATAFORMAT_SUBTYPE_PCM};
    format->Format.nSamplesPerSec = (DWORD)sampleRate;
    format->Format.nChannels = (WORD) channels;
    /* do not support useless padding, like 24-bit samples stored in 32-bit containers */
    format->Format.wBitsPerSample = (WORD) ((bits + 7) & 0xFFF8);

    if (channels <= 2 && bits <= 16) {
        format->Format.wFormatTag = WAVE_FORMAT_PCM;
        format->Format.cbSize = 0;
    } else {
        format->Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
        format->Format.cbSize = 22;
        format->Samples.wValidBitsPerSample = bits;
        /* no way to specify speaker locations */
        format->dwChannelMask = 0xFFFFFFFF;
        format->SubFormat = subtypePCM;
    }
    format->Format.nBlockAlign = (WORD)((format->Format.wBitsPerSample * format->Format.nChannels) / 8);
    format->Format.nAvgBytesPerSec = format->Format.nSamplesPerSec * format->Format.nBlockAlign;
}

/* fill buffer with silence
 */
void DS_clearBuffer(DS_Info* info, BOOL fromWritePos) {
    UBYTE* pb1=NULL, *pb2=NULL;
    DWORD  cb1=0, cb2=0;
    DWORD flags = 0;
    int start, count;
    TRACE1("> DS_clearBuffer for device %d\n", info->deviceID);
    if (info->isSource)  {
        if (fromWritePos) {
                DWORD playCursor, writeCursor;
                int end;
                if (FAILED(info->playBuffer->GetCurrentPosition(&playCursor, &writeCursor))) {
                    ERROR0("  DS_clearBuffer: ERROR: Failed to get current position.");
                    TRACE0("< DS_clearbuffer\n");
                    return;
                }
                DEBUG_SILENCING2("  DS_clearBuffer: DS playPos=%d  myWritePos=%d", (int) playCursor, (int) info->writePos);
                if (info->writePos >= 0) {
                    start = info->writePos + info->silencedBytes;
                } else {
                    start = writeCursor + info->silencedBytes;
                    //flags |= DSBLOCK_FROMWRITECURSOR;
                }
                while (start >= info->dsBufferSizeInBytes) {
                    start -= info->dsBufferSizeInBytes;
                }

                // fix for bug 6251460 (REGRESSION: short sounds do not play)
                // for unknown reason with hardware DS buffer playCursor sometimes
                // jumps back for little interval (mostly 2-8 bytes) (writeCursor moves forward as usual)
                // The issue happens right after start playing and for short sounds only (less then DS buffer,
                // when whole sound written into the buffer and remaining space filled by silence)
                // the case doesn't produce any audible aftifacts so just catch it to prevent filling
                // whole buffer by silence.
                if (((int)playCursor <= start && start < (int)writeCursor)
                    || (writeCursor < playCursor    // buffer bound is between playCursor & writeCursor
                        && (start < (int)writeCursor || (int)playCursor <= start))) {
                    return;
                }

                count = info->dsBufferSizeInBytes - info->silencedBytes;
                // why / 4?
                //if (count > info->dsBufferSizeInBytes / 4) {
                //    count = info->dsBufferSizeInBytes / 4;
                //}
                end = start + count;
                if ((int) playCursor < start) {
                    playCursor += (DWORD) info->dsBufferSizeInBytes;
                }
                if (start <= (int) playCursor && end > (int) playCursor) {
                    /* at maximum, silence until play cursor */
                    count = (int) playCursor - start;
#ifdef USE_TRACE
                    if ((int) playCursor >= info->dsBufferSizeInBytes) playCursor -= (DWORD) info->dsBufferSizeInBytes;
                    TRACE3("\n  DS_clearBuffer: Start Writing from %d, "
                           "would overwrite playCursor=%d, so reduce count to %d\n",
                           start, playCursor, count);
#endif
                }
                DEBUG_SILENCING2("  clearing buffer from %d, count=%d. ", (int)start, (int) count);
                if (count <= 0) {
                    DEBUG_SILENCING0("\n");
                    TRACE1("< DS_clearBuffer: no need to clear, silencedBytes=%d\n", info->silencedBytes);
                    return;
                }
        } else {
                start = 0;
                count = info->dsBufferSizeInBytes;
                flags |= DSBLOCK_ENTIREBUFFER;
        }
        if (FAILED(info->playBuffer->Lock(start,
                                          count,
                                          (LPVOID*) &pb1, &cb1,
                                          (LPVOID*) &pb2, &cb2, flags))) {
            ERROR0("\n  DS_clearBuffer: ERROR: Failed to lock sound buffer.\n");
            TRACE0("< DS_clearbuffer\n");
            return;
        }
    } else {
        if (FAILED(info->captureBuffer->Lock(0,
                                             info->dsBufferSizeInBytes,
                                             (LPVOID*) &pb1, &cb1,
                                             (LPVOID*) &pb2, &cb2, DSCBLOCK_ENTIREBUFFER))) {
            ERROR0("  DS_clearBuffer: ERROR: Failed to lock sound buffer.\n");
            TRACE0("< DS_clearbuffer\n");
            return;
        }
    }
    if (pb1!=NULL) {
        memset(pb1, (info->bitsPerSample == 8)?128:0, cb1);
    }
    if (pb2!=NULL) {
        memset(pb2, (info->bitsPerSample == 8)?128:0, cb2);
    }
    if (info->isSource)  {
        info->playBuffer->Unlock( pb1, cb1, pb2, cb2 );
        if (!fromWritePos) {
            /* doesn't matter where to start writing next time */
            info->writePos = -1;
            info->silencedBytes = info->dsBufferSizeInBytes;
        } else {
            info->silencedBytes += (cb1+cb2);
            if (info->silencedBytes > info->dsBufferSizeInBytes) {
                ERROR1("  DS_clearbuffer: ERROR: silencedBytes=%d exceeds buffer size!\n",
                       info->silencedBytes);
                info->silencedBytes = info->dsBufferSizeInBytes;
            }
        }
        DEBUG_SILENCING2("  silencedBytes=%d, my writePos=%d\n", (int)info->silencedBytes, (int)info->writePos);
    } else {
        info->captureBuffer->Unlock( pb1, cb1, pb2, cb2 );
    }
    TRACE0("< DS_clearbuffer\n");
}

/* returns pointer to buffer */
void* DS_createSoundBuffer(DS_Info* info,
                          float sampleRate,
                          int sampleSizeInBits,
                          int channels,
                          int bufferSizeInBytes) {
    DSBUFFERDESC dsbdesc;
    DSCBUFFERDESC dscbdesc;
    HRESULT res;
    WAVEFORMATEXTENSIBLE format;
    void* buffer;

    TRACE1("Creating secondary buffer for device %d\n", info->deviceID);
    createWaveFormat(&format,
                     (int) sampleRate,
                     channels,
                     info->frameSize / channels * 8,
                     sampleSizeInBits);

    /* 2 second secondary buffer */
    info->dsBufferSizeInBytes = 2 * ((int) sampleRate) * info->frameSize;

    if (bufferSizeInBytes > info->dsBufferSizeInBytes / 2) {
        bufferSizeInBytes = info->dsBufferSizeInBytes / 2;
    }
    bufferSizeInBytes = (bufferSizeInBytes / info->frameSize) * info->frameSize;
    info->bufferSizeInBytes = bufferSizeInBytes;

    if (info->isSource) {
        memset(&dsbdesc, 0, sizeof(DSBUFFERDESC));
        dsbdesc.dwSize = sizeof(DSBUFFERDESC);
        dsbdesc.dwFlags = DSBCAPS_GETCURRENTPOSITION2
                    | DSBCAPS_GLOBALFOCUS;

        dsbdesc.dwBufferBytes = info->dsBufferSizeInBytes;
        dsbdesc.lpwfxFormat = (WAVEFORMATEX*) &format;
        res = DEV_PLAY(info->deviceID)->CreateSoundBuffer
            (&dsbdesc, (LPDIRECTSOUNDBUFFER*) &buffer, NULL);
    } else {
        memset(&dscbdesc, 0, sizeof(DSCBUFFERDESC));
        dscbdesc.dwSize = sizeof(DSCBUFFERDESC);
        dscbdesc.dwFlags = 0;
        dscbdesc.dwBufferBytes = info->dsBufferSizeInBytes;
        dscbdesc.lpwfxFormat = (WAVEFORMATEX*) &format;
        res = DEV_CAPTURE(info->deviceID)->CreateCaptureBuffer
            (&dscbdesc, (LPDIRECTSOUNDCAPTUREBUFFER*) &buffer, NULL);
    }
    if (FAILED(res)) {
        ERROR1("DS_createSoundBuffer: ERROR: Failed to create sound buffer: %s", TranslateDSError(res));
        return NULL;
    }
    return buffer;
}

void DS_destroySoundBuffer(DS_Info* info) {
    if (info->playBuffer != NULL) {
        info->playBuffer->Release();
        info->playBuffer = NULL;
    }
    if (info->captureBuffer != NULL) {
        info->captureBuffer->Release();
        info->captureBuffer = NULL;
    }
}


void* DAUDIO_Open(INT32 mixerIndex, INT32 deviceID, int isSource,
                  int encoding, float sampleRate, int sampleSizeInBits,
                  int frameSize, int channels,
                  int isSigned, int isBigEndian, int bufferSizeInBytes) {

    DS_Info* info;
    void* buffer;

    TRACE0("> DAUDIO_Open\n");

    /* some sanity checks */
    if (deviceID >= g_cacheCount) {
        ERROR1("DAUDIO_Open: ERROR: cannot open the device with deviceID=%d!\n", deviceID);
        return NULL;
    }
    if ((g_audioDeviceCache[deviceID].isSource && !isSource)
        || (!g_audioDeviceCache[deviceID].isSource && isSource)) {
        /* only support Playback or Capture */
        ERROR0("DAUDIO_Open: ERROR: Cache is corrupt: cannot open the device in specified isSource mode!\n");
        return NULL;
    }
    if (encoding != DAUDIO_PCM) {
        ERROR1("DAUDIO_Open: ERROR: cannot open the device with encoding=%d!\n", encoding);
        return NULL;
    }
    if (channels <= 0) {
        ERROR1("DAUDIO_Open: ERROR: Invalid number of channels=%d!\n", channels);
        return NULL;
    }
    if (sampleSizeInBits > 8 &&
#ifdef _LITTLE_ENDIAN
        isBigEndian
#else
        !isBigEndian
#endif
        ) {
        ERROR1("DAUDIO_Open: ERROR: wrong endianness: isBigEndian==%d!\n", isBigEndian);
        return NULL;
    }
    if (sampleSizeInBits == 8 && isSigned) {
        ERROR0("DAUDIO_Open: ERROR: wrong signed'ness: with 8 bits, data must be unsigned!\n");
        return NULL;
    }
    if (!DS_StartBufferHelper::isInitialized()) {
        ERROR0("DAUDIO_Open: ERROR: StartBufferHelper initialization was failed!\n");
        return NULL;
    }

    info = (DS_Info*) malloc(sizeof(DS_Info));
    if (!info) {
        ERROR0("DAUDIO_Open: ERROR: Out of memory\n");
        return NULL;
    }
    memset(info, 0, sizeof(DS_Info));

    info->deviceID = deviceID;
    info->isSource = isSource;
    info->bitsPerSample = sampleSizeInBits;
    info->frameSize = frameSize;
    info->framePos = 0;
    info->started = FALSE;
    info->underrun = FALSE;

    if (!DS_addDeviceRef(deviceID)) {
        DS_removeDeviceRef(deviceID);
        free(info);
        return NULL;
    }

    buffer = DS_createSoundBuffer(info,
                                  sampleRate,
                                  sampleSizeInBits,
                                  channels,
                                  bufferSizeInBytes);
    if (!buffer) {
        DS_removeDeviceRef(deviceID);
        free(info);
        return NULL;
    }

    if (info->isSource) {
        info->playBuffer = (LPDIRECTSOUNDBUFFER) buffer;
    } else {
        info->captureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) buffer;
    }
    DS_clearBuffer(info, FALSE /* entire buffer */);

    /* use writepos of device */
    if (info->isSource) {
        info->writePos = -1;
    } else {
        info->writePos = 0;
    }

    TRACE0("< DAUDIO_Open: Opened device successfully.\n");
    return (void*) info;
}

int DAUDIO_Start(void* id, int isSource) {
    DS_Info* info = (DS_Info*) id;
    HRESULT res = DS_OK;
    DWORD status;

    TRACE0("> DAUDIO_Start\n");

    if (info->isSource)  {
        res = info->playBuffer->GetStatus(&status);
        if (res == DS_OK) {
            if (status & DSBSTATUS_LOOPING) {
                ERROR0("DAUDIO_Start: ERROR: Already started!");
                return TRUE;
            }

            /* only start buffer if already something written to it */
            if (info->writePos >= 0) {
                res = DS_StartBufferHelper::StartBuffer(info);
                if (res == DSERR_BUFFERLOST) {
                    res = info->playBuffer->Restore();
                    if (res == DS_OK) {
                        DS_clearBuffer(info, FALSE /* entire buffer */);
                        /* write() will trigger actual device start */
                    }
                } else {
                    /* make sure that we will have silence after
                       the currently valid audio data */
                    DS_clearBuffer(info, TRUE /* from write position */);
                }
            }
        }
    } else {
        if (info->captureBuffer->GetStatus(&status) == DS_OK) {
            if (status & DSCBSTATUS_LOOPING) {
                ERROR0("DAUDIO_Start: ERROR: Already started!");
                return TRUE;
            }
        }
        res = DS_StartBufferHelper::StartBuffer(info);
    }
    if (FAILED(res)) {
        ERROR1("DAUDIO_Start: ERROR: Failed to start: %s", TranslateDSError(res));
        return FALSE;
    }
    info->started = TRUE;
    return TRUE;
}

int DAUDIO_Stop(void* id, int isSource) {
    DS_Info* info = (DS_Info*) id;

    TRACE0("> DAUDIO_Stop\n");

    info->started = FALSE;
    if (info->isSource)  {
        info->playBuffer->Stop();
    } else {
        info->captureBuffer->Stop();
    }

    TRACE0("< DAUDIO_Stop\n");
    return TRUE;
}


void DAUDIO_Close(void* id, int isSource) {
    DS_Info* info = (DS_Info*) id;

    TRACE0("DAUDIO_Close\n");

    if (info != NULL) {
        DS_destroySoundBuffer(info);
        DS_removeDeviceRef(info->deviceID);
        free(info);
    }
}

/* Check buffer for underrun
 * This method is only meaningful for Output devices (write devices).
 */
void DS_CheckUnderrun(DS_Info* info, DWORD playCursor, DWORD writeCursor) {
    TRACE5("DS_CheckUnderrun: playCursor=%d, writeCursor=%d, "
           "info->writePos=%d  silencedBytes=%d  dsBufferSizeInBytes=%d\n",
           (int) playCursor, (int) writeCursor, (int) info->writePos,
           (int) info->silencedBytes, (int) info->dsBufferSizeInBytes);
    if (info->underrun || info->writePos < 0) return;
    int writeAhead = DS_getDistance(info, writeCursor, info->writePos);
    if (writeAhead > info->bufferSizeInBytes) {
        // this may occur after Stop(), when writeCursor decreases (real valid data size > bufferSizeInBytes)
        // But the case can occur only when we have more then info->bufferSizeInBytes valid bytes
        // (and less then (info->dsBufferSizeInBytes - info->bufferSizeInBytes) silenced bytes)
        // If we already have a lot of silencedBytes after valid data (written by
        // DAUDIO_StillDraining() or DAUDIO_Service()) then it's underrun
        if (info->silencedBytes >= info->dsBufferSizeInBytes - info->bufferSizeInBytes) {
            // underrun!
            ERROR0("DS_CheckUnderrun: ERROR: underrun detected!\n");
            info->underrun = TRUE;
        }
    }
}

/* For source (playback) line:
 *   (a) if (fromPlayCursor == FALSE), returns number of bytes available
 *     for writing: bufferSize - (info->writePos - writeCursor);
 *   (b) if (fromPlayCursor == TRUE), playCursor is used instead writeCursor
 *     and returned value can be used for play position calculation (see also
 *     note about bufferSize)
 * For destination (capture) line:
 *   (c) if (fromPlayCursor == FALSE), returns number of bytes available
 *     for reading from the buffer: readCursor - info->writePos;
 *   (d) if (fromPlayCursor == TRUE), captureCursor is used instead readCursor
 *     and returned value can be used for capture position calculation (see
 *     note about bufferSize)
 * bufferSize parameter are filled by "actual" buffer size:
 *   if (fromPlayCursor == FALSE), bufferSize = info->bufferSizeInBytes
 *   otherwise it increase by number of bytes currently processed by DirectSound
 *     (writeCursor - playCursor) or (captureCursor - readCursor)
 */
int DS_GetAvailable(DS_Info* info,
                    DWORD* playCursor, DWORD* writeCursor,
                    int* bufferSize, BOOL fromPlayCursor) {
    int available;
    int newReadPos;

    TRACE2("DS_GetAvailable: fromPlayCursor=%d,  deviceID=%d\n", fromPlayCursor, info->deviceID);
    if (!info->playBuffer && !info->captureBuffer) {
        ERROR0("DS_GetAvailable: ERROR: buffer not yet created");
        return 0;
    }

    if (info->isSource)  {
        if (FAILED(info->playBuffer->GetCurrentPosition(playCursor, writeCursor))) {
            ERROR0("DS_GetAvailable: ERROR: Failed to get current position.\n");
            return 0;
        }
        int processing = DS_getDistance(info, (int)*playCursor, (int)*writeCursor);
        // workaround: sometimes DirectSound report writeCursor is less (for several bytes) then playCursor
        if (processing > info->dsBufferSizeInBytes / 2) {
            *writeCursor = *playCursor;
            processing = 0;
        }
        TRACE3("   playCursor=%d, writeCursor=%d, info->writePos=%d\n",
               *playCursor, *writeCursor, info->writePos);
        *bufferSize = info->bufferSizeInBytes;
        if (fromPlayCursor) {
            *bufferSize += processing;
        }
        DS_CheckUnderrun(info, *playCursor, *writeCursor);
        if (info->writePos == -1 || (info->underrun && !fromPlayCursor)) {
                /* always full buffer if at beginning */
                available = *bufferSize;
        } else {
            int currWriteAhead = DS_getDistance(info, fromPlayCursor ? (int)*playCursor : (int)*writeCursor, info->writePos);
            if (currWriteAhead > *bufferSize) {
                if (info->underrun) {
                    // playCursor surpassed writePos - no valid data, whole buffer available
                    available = *bufferSize;
                } else {
                    // the case may occur after stop(), when writeCursor jumps back to playCursor
                    // so "actual" buffer size has grown
                    *bufferSize = currWriteAhead;
                    available = 0;
                }
            } else {
                available = *bufferSize - currWriteAhead;
            }
        }
    } else {
        if (FAILED(info->captureBuffer->GetCurrentPosition(playCursor, writeCursor))) {
            ERROR0("DS_GetAvailable: ERROR: Failed to get current position.\n");
            return 0;
        }
        *bufferSize = info->bufferSizeInBytes;
        if (fromPlayCursor) {
            *bufferSize += DS_getDistance(info, (int)*playCursor, (int)*writeCursor);
        }
        TRACE4("   captureCursor=%d, readCursor=%d, info->readPos=%d  refBufferSize=%d\n",
               *playCursor, *writeCursor, info->writePos, *bufferSize);
        if (info->writePos == -1) {
            /* always empty buffer if at beginning */
            info->writePos = (int) (*writeCursor);
        }
        if (fromPlayCursor) {
            available = ((int) (*playCursor) - info->writePos);
        } else {
            available = ((int) (*writeCursor) - info->writePos);
        }
        if (available < 0) {
            available += info->dsBufferSizeInBytes;
        }
        if (!fromPlayCursor && available > info->bufferSizeInBytes) {
            /* overflow */
            ERROR2("DS_GetAvailable: ERROR: overflow detected: "
                   "DirectSoundBufferSize=%d, bufferSize=%d, ",
                   info->dsBufferSizeInBytes, info->bufferSizeInBytes);
            ERROR3("captureCursor=%d, readCursor=%d, info->readPos=%d\n",
                   *playCursor, *writeCursor, info->writePos);
            /* advance read position, to allow exactly one buffer worth of data */
            newReadPos = (int) (*writeCursor) - info->bufferSizeInBytes;
            if (newReadPos < 0) {
                newReadPos += info->dsBufferSizeInBytes;
            }
            info->writePos = newReadPos;
            available = info->bufferSizeInBytes;
        }
    }
    available = (available / info->frameSize) * info->frameSize;

    TRACE1("DS_available: Returning %d available bytes\n", (int) available);
    return available;
}

// returns -1 on error, otherwise bytes written
int DAUDIO_Write(void* id, char* data, int byteSize) {
    DS_Info* info = (DS_Info*) id;
    int available;
    int thisWritePos;
    DWORD playCursor, writeCursor;
    HRESULT res;
    void* buffer1, *buffer2;
    DWORD buffer1len, buffer2len;
    BOOL needRestart = FALSE;
    int bufferLostTrials = 2;
    int bufferSize;

    TRACE1("> DAUDIO_Write %d bytes\n", byteSize);

    while (--bufferLostTrials > 0) {
        available = DS_GetAvailable(info, &playCursor, &writeCursor, &bufferSize, FALSE /* fromPlayCursor */);
        if (byteSize > available) byteSize = available;
        if (byteSize == 0) break;
        thisWritePos = info->writePos;
        if (thisWritePos == -1 || info->underrun) {
            // play from current write cursor after flush, etc.
            needRestart = TRUE;
            thisWritePos = writeCursor;
            info->underrun = FALSE;
        }
        DEBUG_SILENCING2("DAUDIO_Write: writing from %d, count=%d\n", (int) thisWritePos, (int) byteSize);
        res = info->playBuffer->Lock(thisWritePos, byteSize,
                                     (LPVOID *) &buffer1, &buffer1len,
                                     (LPVOID *) &buffer2, &buffer2len,
                                     0);
        if (res != DS_OK) {
            /* some DS failure */
            if (res == DSERR_BUFFERLOST) {
                ERROR0("DAUDIO_write: ERROR: Restoring lost Buffer.");
                if (info->playBuffer->Restore() == DS_OK) {
                    DS_clearBuffer(info, FALSE /* entire buffer */);
                    info->writePos = -1;
                    /* try again */
                    continue;
                }
            }
            /* can't recover from error */
            byteSize = 0;
            break;
        }
        /* buffer could be locked successfully */
        /* first fill first buffer */
        if (buffer1) {
            memcpy(buffer1, data, buffer1len);
            data = (char*) (((UINT_PTR) data) + buffer1len);
        } else buffer1len = 0;
        if (buffer2) {
            memcpy(buffer2, data, buffer2len);
        } else buffer2len = 0;
        byteSize = buffer1len + buffer2len;

        /* update next write pos */
        thisWritePos += byteSize;
        while (thisWritePos >= info->dsBufferSizeInBytes) {
            thisWritePos -= info->dsBufferSizeInBytes;
        }
        /* commit data to directsound */
        info->playBuffer->Unlock(buffer1, buffer1len, buffer2, buffer2len);

        info->writePos = thisWritePos;

        /* update position
         * must be AFTER updating writePos,
         * so that getSvailable doesn't return too little,
         * so that getFramePos doesn't jump
         */
        info->framePos += (byteSize / info->frameSize);

        /* decrease silenced bytes */
        if (info->silencedBytes > byteSize) {
            info->silencedBytes -= byteSize;
        } else {
            info->silencedBytes = 0;
        }
        break;
    } /* while */

    /* start the device, if necessary */
    if (info->started && needRestart && (info->writePos >= 0)) {
        DS_StartBufferHelper::StartBuffer(info);
    }

    TRACE1("< DAUDIO_Write: returning %d bytes.\n", byteSize);
    return byteSize;
}

// returns -1 on error
int DAUDIO_Read(void* id, char* data, int byteSize) {
    DS_Info* info = (DS_Info*) id;
    int available;
    int thisReadPos;
    DWORD captureCursor, readCursor;
    HRESULT res;
    void* buffer1, *buffer2;
    DWORD buffer1len, buffer2len;
    int bufferSize;

    TRACE1("> DAUDIO_Read %d bytes\n", byteSize);

    available = DS_GetAvailable(info, &captureCursor, &readCursor, &bufferSize, FALSE /* fromCaptureCursor? */);
    if (byteSize > available) byteSize = available;
    if (byteSize > 0) {
        thisReadPos = info->writePos;
        if (thisReadPos == -1) {
            /* from beginning */
            thisReadPos = 0;
        }
        res = info->captureBuffer->Lock(thisReadPos, byteSize,
                                        (LPVOID *) &buffer1, &buffer1len,
                                        (LPVOID *) &buffer2, &buffer2len,
                                        0);
        if (res != DS_OK) {
            /* can't recover from error */
            byteSize = 0;
        } else {
            /* buffer could be locked successfully */
            /* first fill first buffer */
            if (buffer1) {
                memcpy(data, buffer1, buffer1len);
                data = (char*) (((UINT_PTR) data) + buffer1len);
            } else buffer1len = 0;
            if (buffer2) {
                memcpy(data, buffer2, buffer2len);
            } else buffer2len = 0;
            byteSize = buffer1len + buffer2len;

            /* update next read pos */
            thisReadPos = DS_addPos(info, thisReadPos, byteSize);
            /* commit data to directsound */
            info->captureBuffer->Unlock(buffer1, buffer1len, buffer2, buffer2len);

            /* update position
             * must be BEFORE updating readPos,
             * so that getAvailable doesn't return too much,
             * so that getFramePos doesn't jump
             */
            info->framePos += (byteSize / info->frameSize);

            info->writePos = thisReadPos;
        }
    }

    TRACE1("< DAUDIO_Read: returning %d bytes.\n", byteSize);
    return byteSize;
}


int DAUDIO_GetBufferSize(void* id, int isSource) {
    DS_Info* info = (DS_Info*) id;
    return info->bufferSizeInBytes;
}

int DAUDIO_StillDraining(void* id, int isSource) {
    DS_Info* info = (DS_Info*) id;
    BOOL draining = FALSE;
    int available, bufferSize;
    DWORD playCursor, writeCursor;

    DS_clearBuffer(info, TRUE /* from write position */);
    available = DS_GetAvailable(info, &playCursor, &writeCursor, &bufferSize, TRUE /* fromPlayCursor */);
    draining = (available < bufferSize);

    TRACE3("DAUDIO_StillDraining: available=%d  silencedBytes=%d  Still draining: %s\n",
           available, info->silencedBytes, draining?"TRUE":"FALSE");
    return draining;
}


int DAUDIO_Flush(void* id, int isSource) {
    DS_Info* info = (DS_Info*) id;

    TRACE0("DAUDIO_Flush\n");

    if (info->isSource)  {
        info->playBuffer->Stop();
        DS_clearBuffer(info, FALSE /* entire buffer */);
    } else {
        DWORD captureCursor, readCursor;
        /* set the read pointer to the current read position */
        if (FAILED(info->captureBuffer->GetCurrentPosition(&captureCursor, &readCursor))) {
            ERROR0("DAUDIO_Flush: ERROR: Failed to get current position.");
            return FALSE;
        }
        DS_clearBuffer(info, FALSE /* entire buffer */);
        /* SHOULD set to *captureCursor*,
         * but that would be detected as overflow
         * in a subsequent GetAvailable() call.
         */
        info->writePos = (int) readCursor;
    }
    return TRUE;
}

int DAUDIO_GetAvailable(void* id, int isSource) {
    DS_Info* info = (DS_Info*) id;
    DWORD playCursor, writeCursor;
    int ret, bufferSize;

    ret = DS_GetAvailable(info, &playCursor, &writeCursor, &bufferSize, /*fromPlayCursor?*/ FALSE);

    TRACE1("DAUDIO_GetAvailable returns %d bytes\n", ret);
    return ret;
}

INT64 estimatePositionFromAvail(DS_Info* info, INT64 javaBytePos, int bufferSize, int availInBytes) {
    // estimate the current position with the buffer size and
    // the available bytes to read or write in the buffer.
    // not an elegant solution - bytePos will stop on xruns,
    // and in race conditions it may jump backwards
    // Advantage is that it is indeed based on the samples that go through
    // the system (rather than time-based methods)
    if (info->isSource) {
        // javaBytePos is the position that is reached when the current
        // buffer is played completely
        return (INT64) (javaBytePos - bufferSize + availInBytes);
    } else {
        // javaBytePos is the position that was when the current buffer was empty
        return (INT64) (javaBytePos + availInBytes);
    }
}

INT64 DAUDIO_GetBytePosition(void* id, int isSource, INT64 javaBytePos) {
    DS_Info* info = (DS_Info*) id;
    int available, bufferSize;
    DWORD playCursor, writeCursor;
    INT64 result = javaBytePos;

    available = DS_GetAvailable(info, &playCursor, &writeCursor, &bufferSize, /*fromPlayCursor?*/ TRUE);
    result = estimatePositionFromAvail(info, javaBytePos, bufferSize, available);
    return result;
}


void DAUDIO_SetBytePosition(void* id, int isSource, INT64 javaBytePos) {
    /* save to ignore, since GetBytePosition
     * takes the javaBytePos param into account
     */
}

int DAUDIO_RequiresServicing(void* id, int isSource) {
    // need servicing on for SourceDataLines
    return isSource?TRUE:FALSE;
}

void DAUDIO_Service(void* id, int isSource) {
    DS_Info* info = (DS_Info*) id;
    if (isSource) {
        if (info->silencedBytes < info->dsBufferSizeInBytes) {
            // clear buffer
            TRACE0("DAUDIO_Service\n");
            DS_clearBuffer(info, TRUE /* from write position */);
        }
        if (info->writePos >= 0
            && info->started
            && !info->underrun
            && info->silencedBytes >= info->dsBufferSizeInBytes) {
            // if we're currently playing, and the entire buffer is silenced...
            // then we are underrunning!
            info->underrun = TRUE;
            ERROR0("DAUDIO_Service: ERROR: DirectSound: underrun detected!\n");
        }
    }
}


#endif // USE_DAUDIO